I can't see any reason it shouldn't be.
At this stage, after reading for the past couple of days, my two main
concerns are NAT handling of SIP as both the Asterisk & my clients will
be behind a firewall on a private net, and multitasking - the latter
*may* be solved by going with AGI (not sure yet as Asterisk is still
completely new to me).
I figured out most of the things which had me worried initially,
including how to get multiple "register" entries for external providers
using (non-standard) ports (in v10.0 there is a provision for this in
sip.conf).
If so, I am not completely clear on whether I need to explicitly specify
my public IP address (via externip/externhost) or whether Asterick is
able to find it without this option?
As I understand it, that depends on your router. If you have a Linux
router with the ip_nat_sip module, it'll "fix" your SIP packets so
that you don't need to use the externip setting. However, you'll need
to test to verify that.
Nope! My eth0 interface is not facing the public Internet directly - it
takes its IP address from my ISP's DHCP (which is private!) even though
it can forward/pass traffic through the public internet via that
interface, that is the problem.
Asterisk won't be able to figure out your external address on its own,
so if your firewall isn't fixing packets, then you'd need to specify
externip.
I had a brief look at the sip.conf(.sample) for v10.0 and there is a
provision for activating STUN (application/module) to figure out what my
"real" public address is - if it works, then I may as well go using
this, otherwise I will have to use a separate program to do that job.
http://www.voip-info.org/wiki/view/Asterisk+variables
According to the information here, you should be able to use
${ENV(externip)} to reference the value of an environment variable
named "externip".
Thanks, that was good, but this is better ;-) ->
http://www.voip-info.org/wiki/view/Asterisk+Detailed+Variable+List
http://www.asteriskguru.com/tutorials/sip_nat_oneway_or_no_audio_asterisk.html
For a SIP trunk... no, I don't. The above link may be useful as it
describes NAT issues with SIP. If you have to specify NAT options at
all, start with "yes" and try "route" if that doesn't work.
Very good find, thanks again!
Is there a comprehensive list of all the options available in sip.conf
and what they do, because I was unable to find such a list?
http://www.voip-info.org/wiki/view/Asterisk+config+sip.conf
I wish I knew. The link above seems fairly complete, but also terse.
Top find again, thanks! It is a bit dated, but it certainly helps and
I've got a few ideas of my own from this page.
One final question about binding: in order to be able to use both tun0
and eth1 interfaces so that Asterick serves the calls from both eth1 and
tun0, do I have to use "bind 0.0.0.0"? Is there an alternative, like
specifying "bind 10.1.1.1" for eth1 and then "bind 10.1.2.1" for the
tun0 interface - is this possible?
Start with binding to 0.0.0.0.
That was my initial intention as I was hoping Linux will map each
request/response using the appropriate interface (i.e. on which
interface it comes from), I realise binding on 0.0.0.0. is not ideal
from a security point of view (I'd rather issue separate udpbind
statements for the interfaces I want to use), but for now it have to do
if there isn't an alternative.
Many thanks for your input, much appreciated.
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